[Freeswitch-users] One way audio to CME
Anthony McGarry
agtmcgarry at gmail.com
Thu Aug 8 13:07:46 MSD 2013
sip-ip & rtp-ip should be set per profile. So if you have 5 profiles each profile.xml will have these params set to whatever IP address the profile should use.
In your case
external profile
<param name="sip-ip" value="10.1.1.206"/>
<param name="rtp-ip" value="10.1.1.206"/>
internal profile
<param name="sip-ip" value="10.10.10.206"/>
<param name="rtp-ip" value="10.10.10.206"/>
you only need to worry about ext-sip-ip and ext-rtp-ip if there is NAT in the path.
Better to leave them in each profile to
<param name="ext-sip-ip" value="auto-nat"/>
<param name="ext-rtp-ip" value="auto-nat"/>
On 8 Aug 2013, at 03:29, Peter <eidevm5 at gmail.com> wrote:
> Hi Brian.
>
> I did have the IP set in the external profile, but after removing them it didn't seem to make any difference, ie: the SIP ports were listening on the correct IP address.
>
> Have to say I'm getting a little confused as to when and where to use sip-ip and ext-sip-ip
>
> Should sip-ip ALWAYS be set to your internal IP address and ext-sip-ip to your external IP address?
>
> If so, which profiles do they need to be set in? If they are set in both the internal and external SIP profiles, does that matter?
>
> Thanks
>
> Peter
>
>
>
> On Thu, Aug 8, 2013 at 11:41 AM, Brian Foster <bdfoster at davri.com> wrote:
> Your ip for the external profile will show whatever the ip is set to in the external profile configs. It doesn't matter what is set in the internal profile.
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana
>
> Sent from a mobile device.
>
> On Aug 7, 2013 9:36 PM, "Peter" <eidevm5 at gmail.com> wrote:
> Hi Anthony
>
> Really appreciate the taking your time to look at this. It's starting to drive me nuts.
>
> My dialplan for calls to CME is:
>
> <extension name="incoming-cme">
> <condition field="destination_number" expression="^(3\d\d\d)$">
> <action application="bridge" data="sofia/internal/${destination_number}@10.10.10.203"/>
> </condition>
> </extension>
>
>
> My internal sip profile has:
>
> <param name="rtp-ip" value="10.10.10.206"/>
> <param name="sip-ip" value="10.10.10.206"/>
> <param name="ext-rtp-ip" value="10.1.1.206"/>
> <param name="ext-sip-ip" value="10.1.1.206"/>
>
> The output from:
>
> sofia status
>
> is:
>
> internal profile sip:mod_sofia at 10.1.1.206:5060 RUNNING (0)
> external profile sip:mod_sofia at 10.1.1.206:5060 RUNNING (0)
>
> Should internal show 10.10.10.206??
>
> The output from:
>
> sofia status profile internal
>
> shows:
>
> Name internal
> Domain Name N/A
> Auto-NAT false
> DBName sofia_reg_internal
> Pres Hosts 10.10.10.206,10.1.1.206
> Dialplan XML
> Context public
> Challenge Realm auto_from
> RTP-IP 10.10.10.206
> Ext-RTP-IP 10.1.1.206
> SIP-IP 10.10.10.206
> Ext-SIP-IP 10.1.1.206
> URL sip:mod_sofia at 10.1.1.206:5060
> BIND-URL sip:mod_sofia at 10.1.1.206:5060;maddr=10.10.10.206
> HOLD-MUSIC N/A
> OUTBOUND-PROXY N/A
> CODECS IN iLBC at 30i,PCMU,PCMA,GSM
> CODECS OUT iLBC at 30i,PCMU,PCMA,GSM
> TEL-EVENT 101
> DTMF-MODE rfc2833
> CNG 13
> SESSION-TO 0
> MAX-DIALOG 0
> NOMEDIA false
> LATE-NEG false
> PROXY-MEDIA false
> ZRTP-PASSTHRU false
> AGGRESSIVENAT false
> STUN-ENABLED true
> STUN-AUTO-DISABLE false
>
>
> The SIP trace from the Freeswitch SBC is at:
>
> http://pastebin.freeswitch.org/21279
>
> I've been playing around with all sorts of different combinations of SIP/RTP IP settings, but still no closer.
>
> Hope you have some insight.
>
> Thanks
>
> Peter
>
> On Wed, Aug 7, 2013 at 6:31 PM, Anthony McGarry <agtmcgarry at gmail.com> wrote:
> Hi Peter,
>
> Your debug shows the invite with via/from/contact/rpid all coming from 10.1.1.206, your external side.
> Check your bridge statement, is it using the correct sip profile?
> Check your sip profile SBC internal params rtp-ip & sip-ip, make sure they are set correctly to 10.10.10.206
> Paste up your logs from the sbc including sip trace.
>
> Anthony
>
>
> On 7 Aug 2013, at 08:12, Peter <eidevm5 at gmail.com> wrote:
>
>> Hi Anthony.
>>
>> Yes, the SIP profiles are the same for calls going to Kamailio and to CME/CUBE.
>>
>> Note that CME only has one interface, so binding the source interface doesn't really make much sense.
>>
>> Note that I've simplified my set up a little and the phone that was registered to CUCM is now registered to CME. However, the result is still the same, ie: one way audio to the Cisco phone.
>>
>> You can see the SIP debug from CME at:
>>
>> http://pastebin.freeswitch.org/21274
>>
>> The call is coming from 1001 at 10.1.1.204 to 3000 at 10.10.10.203
>>
>> where
>>
>> 10.1.1.204 - Freeswitch where SIP clients register to
>> 10.1.1.206 - External side of Freeswitch SBC
>> 10.10.10.206 - Internal side of Freeswitch SBC
>> 10.10.10.203 - CME
>>
>> Peter
>>
>>
>>
>> On Tue, Aug 6, 2013 at 5:13 PM, Anthony McGarry <agtmcgarry at gmail.com> wrote:
>> Hi Peter,
>>
>> Because the calls are fine when using Kamailio I'm assuming your sip profiles are fine and you FS SBC config is fine. Are you using the same profiles?
>> Yes you are correct. Have you added the commands? Add them as a first step.
>> Send on a 'debug ccsip messages'
>>
>> Anthony
>>
>>
>>
>> On 6 Aug 2013, at 05:35, Peter <eidevm5 at gmail.com> wrote:
>>
>>> Thanks for replying Anthony.
>>>
>>> Keep in mind that I have very little experience with Cisco products, so I may be missing something fundamental.
>>>
>>> As far as I can see
>>>
>>> voice-class sip bind media source-interface ....
>>>
>>> is just used to bind the SIP or media stream to the appropriate interface on the CUBE.
>>>
>>> My issue is that the CUBE is trying to initiate the return RTP stream to the external interface (instead of the internal interface) on the Freeswitch SBC.
>>>
>>> Is my understanding of the sip bind media command correct?
>>>
>>> Thanks
>>>
>>> Peter
>>>
>>>
>>> On Mon, Aug 5, 2013 at 5:23 PM, Anthony McGarry <agtmcgarry at gmail.com> wrote:
>>> On cube make sure you specify the source address on your dial-peers
>>> voice-class sip bind media|control
>>> to the correct side. I have seen one way audio when not set.
>>>
>>> On 5 Aug 2013, at 06:29, Peter <eidevm5 at gmail.com> wrote:
>>>
>>> >
>>> >
>>> > I currently have successful two way calls (signalling and media) in the following setup
>>> >
>>> >
>>> > External Linphone --> Freeswitch --> Freeswitch SBC -> Router -> Kamailio --> Internal Linphone
>>> >
>>> > However, when I try to call a Cisco handset that is registered to CUCM9 via CME in the following config:
>>> >
>>> > External Linphone --> Freeswitch --> Freeswitch SBC -> Router -> CME -> CUCM9 --> Cisco handset
>>> >
>>> > The call signalling appears to be working fine and I can successfully initiate a call from each end, but the only RTP stream that is working is from the external Linphone client to the Cisco handset.
>>> >
>>> > Note that CME is being used as a CUBE device, so all SIP and RTP goes via it.
>>> >
>>> > Looking at the RTP debugs on CME I can see the problem is that the "Media Dest Addr" is getting set to the external side of the FS SBC rather than the internal IP address.
>>> >
>>> >
>>> > I tried setting adding:
>>> >
>>> > <action application="set" data="disable_rtp_auto_adjust="true" />
>>> >
>>> > to the dialplan on the SBC, but it made no difference.
>>> >
>>> >
>>> > Any suggestions as to what to check next?
>>> >
>>> > Thanks
>>> >
>>> > Peter
>>> >
>>> > _________________________________________________________________________
>>>
>>>
>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>>
>>>
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>
>>
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130808/668bc04f/attachment-0001.html
Join us at ClueCon 2013 Aug 6-8, 2013
More information about the FreeSWITCH-users
mailing list